Commit graph

643 commits

Author SHA1 Message Date
François Laignel
fd154d272c webrtcsrc: add do-retransmission property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
François Laignel
ede354d7a5 webrtcsink: prevent video-info error log for audio streams
The following error is logged when `webrtcsink` is feeded with an audio stream:

> ERROR video-info video-info.c:540:gst_video_info_from_caps:
>       wrong name 'audio/x-raw', expected video/ or image/

This commit bypasses `VideoInfo::from_caps` for audio streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
François Laignel
158fe80779 rtp: gccbwe: don't break downstream assumptions pushing buffer lists
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.

`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.

This commit makes `rtpgccbwe` push the buffers as they were received: one by one.

[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
Guillaume Desmottes
a502dba6d5 webrtc: janus: handle 'hangup' messages from Janus
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
Guillaume Desmottes
be055f6dfa webrtc: janus: handle 'destroyed' messages from Janus
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
Sebastian Dröge
c982db73a7 rtp: Switch from chrono to time
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-21 13:30:20 +02:00
Guillaume Desmottes
21f59c65da webrtc: allow resolution and framerate input changes
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
7a90e96332 livekit_signaller: Added missing getter for excluded-producer-peer-ids
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
283d1568b4 webrtcsrc: Removed incorrect URIHandler from LiveKit source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
51287705ce livekit_signaller: Improved shutdown behavior
Without sending a Leave request to the server before disconnecting, the
disconnected client will appear present and stuck in the room for a little
while until the server removes it due to inactivity.

After this change, the disconnecting client will immediately leave the room.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
e9edee131b webrtcsrc: Removed flag setup from WhipServerSrc
It's already done in the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
feb01510f9 webrtcsrc: Updated readme for LiveKit source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
32e13f0a10 webrtcsrc: Added LiveKit source element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
c8dcd50846 webrtcsink: Updated livekitwebrtcsink for new signaller constructor
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Jordan Yelloz
59ee2721bf livekit_signaller: Added dual-role support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:24:55 +01:00
Guillaume Desmottes
133b527391 webrtc: janus: rename RoomId to JanusId
Those weird ids are used in multiple places, not only for the room id,
so best to have a more generic name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:54:45 +02:00
Guillaume Desmottes
7f460c2db8 webrtc: janus: room id not optional in 'joined' message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:54:17 +02:00
Guillaume Desmottes
8361471fcc webrtc: janus: remove 'audio' and 'video' from publish messages
Those are deprecated and no longer used.

See https://janus.conf.meetecho.com/docs/videoroom and
https://github.com/meetecho/janus-gateway/blob/master/src/plugins/janus_videoroom.c#L9894

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:54:02 +02:00
Guillaume Desmottes
b9ea05a14a webrtc: janus: numerical room ids are u64
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:53:51 +02:00
Jordan Yelloz
048d51d9d9 webrtcsrc: Made producer-peer-id optional
It may be necessary for some signalling clients but the source element
doesn't need to depend on it.

Also, the value will fall back to the pad's MSID for the first argument
to the request-encoded-filter gobject signal when it isn't available
from the signalling client.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
2024-03-20 16:53:22 +02:00
Xavier Claessens
7edf94f98b janusvr: Add string-ids property
It forces usage of strings even if it can be parsed into an integer.
This allows joining room `"133"` in a server configured with string
room ids.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:24:29 +02:00
Xavier Claessens
ea59544c71 janusvr: Room IDs can be strings
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:24:25 +02:00
Maksym Khomenko
4e86b0f3c8 webrtcsink: extensions: separate API and signal checks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:24:02 +02:00
Maksym Khomenko
98411e97f1 webrtcsink: apply rustfmt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:23:56 +02:00
Xavier Claessens
3a0f30be96 janusvr: Add secret-key property
Every API calls have an optional "apisecret" argument.

Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:23:45 +02:00
Sebastian Dröge
3679db5740 rtpgccbwe: Don't reset PTS/DTS to None
The element is usually placed before `rtpsession`, and `rtpsession`
needs the PTS/DTS for correctly determining the running time. The
running time is then used to produce correct RTCP SR, and to potentially
update an NTP-64 RTP header extension if existing on the packets.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/496

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
2024-02-26 14:20:53 +02:00
Jordan Yelloz
606352d7cf webrtcsink: Added sinkpad with "msid" property
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>
2024-02-12 18:11:42 +02:00
Sebastian Dröge
aa2d056ea1 Update to async-tungstenite 0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>
2024-02-12 18:11:31 +02:00
Sebastian Dröge
16e001e3f2 Update dependency versions for gtk-rs-core / gtk4-rs / gstreamer-rs and local crates 2024-02-08 19:40:08 +02:00
Sebastian Dröge
92891a61e8 Fix a couple of compiler/clippy warnings with --no-default-features
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1452>
2024-02-08 13:02:55 +02:00
Nirbheek Chauhan
cf5e7f6ed3 rtspsrc2: Add some top-level documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
7a1cd675c2 rtspsrc2: Fix RTCP send/recv in the multicast case
Don't use connect(), since that is incompatible with multicast.
Instead, drop received packets that are from senders we do not want.

Also set multicast loopback = false so we don't receive RTCP RRs from
ourselves and interpret them as RTCP SRs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
e59f3bbe58 rtspsrc2: Increase RTP timeout to 5 seconds, matching rtspsrc
Also fix some logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
3e963e9239 rtspsrc2: Implement NetAddressMeta support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
42425abb69 rtspsrc: Factor out SDP → Caps, parse more attributes
This could be a struct of some kind derived from sdp_types::Media etc,
but this is fine for now.

Adds parsing of framesize, and fallbacks for missing or incomplete
rtpmap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
437326ebfd rtspsrc2: Allocate a buffer pool for UDP RTP data
Control the size with a receive-mtu property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
44e49a06a0 rtspsrc2: Emit EOS if any ssrc gets a BYE packet or times out
This allows us to exit when the live-stream ends.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
975556c06b rtspsrc2: Allow a SETUP response without a Transports header
If we only send a single Transport in the Transports header, then the
server is allowed to omit it in the response. This has some strange
consequences for UDP transport: specifically, we have no idea what
addr/port we will get the packets from.

In those cases, we connect() on the socket when we receive the first
packet, so we can send RTCP RRs, and also so we can ensure that we
ignore data from other addresses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
086ffd7aff New RTSP source plugin with live streaming support
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]

Features:
* Live streaming N audio and N video
  - With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
  - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
  - protocols selection and priority (NEW!)
  - location supports rtsp[ut]://
  - port-start instead of port-range

Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:18 +05:30
Bilal Elmoussaoui
0615a16124 Use workspace features for crates metadata/deps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1446>
2024-02-05 15:34:31 +01:00
Sebastian Dröge
91abc62ad0 webrtcsink: Fix new clippy warning
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:53:20 +02:00
Sebastian Dröge
1a55c70114 Switch git dependencies to explicitly name branch
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:51:36 +02:00
Sebastian Dröge
ffa830ae9b Update for GLib prelude re-organization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1444>
2024-02-03 12:30:15 +02:00
Jordan Yelloz
311fda649f livekit_signaller: Added high-quality layer for video streams
This change addresses a cosmetic issue with livekit, where the
connection quality indicator seen by other users shows bad quality
unless the track is added with a high quality layer. The details of the
layer submitted aren't significant for this purpose.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1443>
2024-02-02 20:57:17 +00:00
Robert Ayrapetyan
916a8b959e doc: add http headers for webrtcsink signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
972b9e5474 doc: add docstrings for signaller object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
7a72b2fc25 webrtcsink-signalling: add headers support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
François Laignel
91bfd0f7c3 webrtc: signallers: attempt to close the ws when an error occurs
This commit discards the early error returns in the send tasks to log the error
and attempt to close the websocket.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
François Laignel
f54d714afd webrtc: only use close() to close websockets
In the signaller clients and servers, the following sequence is used to close
the websocket (in the [send task]):

```rust
    ws_sink.send(WsMessage::Close(None)).await?;
    ws_sink.close().await?;
```

tungstenite's [`WebSocket::close()` doc] states:

> Calling this function is the same as calling `write(Message::Close(..))``

So we might think they are redundant and either could be used for this purpose
(`send()` calls `write()`, then `flush()`).

The result is actually is bit different as `write()` starts by checking the
state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from
the peer via a [call to `do_close()`]). Note that `do_close()` also enqueues a
`Close` frame.

This behaviour is visible from the server's logs:

```
1. tungstenite::protocol: Received close frame: None
2. tungstenite::protocol: Replying to close with Frame { header: FrameHeader { .., opcode: Control(Close), .. }, payload: [] }
3. gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
4. gst_plugin_webrtc_signalling::server: connection closed: None this_id=cb13892f-b4d5-4d59-95e2-b3873a7bd319
5. remove_peer{peer_id="cb13892f-b4d5-4d59-95e2-b3873a7bd319"}: gst_plugin_webrtc_signalling::server: close time.busy=285µs time.idle=55.5µs
6. async_tungstenite: websocket start_send error: WebSocket protocol error: Sending after closing is not allowed
```

1: The server's websocket receives the peer's `Close(None)`.
2: `do_close()` enqueues a `Close` frame.
3: The incoming `Close(None)` is handled by the server.
4 & 5: perform session closing.
6: `ws_sink.send(WsMessage::Close(None))` attempts to `write()` while the ws
   is no longer active. The error causes an early return, which means that
   the enqueued `Close` frame is not flushed.

Depending on the peer's shutdown sequence, this can result in the following
error, which can bubble up as a `Message` on the application's bus:

```
ERROR: from element /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: GStreamer encountered a general stream error.
Additional debug info:
net/webrtc/src/webrtcsrc/imp.rs(625): gstrswebrtc::webrtcsrc:👿:BaseWebRTCSrc::connect_signaller::{{closure}}::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0:
Signalling error: Error receiving: WebSocket protocol error: Connection reset without closing handshake
```

On the other hand, [`close()` ensures the ws is active] before attempting to
write a `Close` frame. If it's not, it only flushes the stream.

Thus, when we want to be able to close the websocket and/or to honor the closing
handshake in response to the peer `Close` message, the `ws_sink.close()`
variant is preferable.

This can be verified in the resulting server's logs:

```
tungstenite::protocol: Received close frame: None
tungstenite::protocol: Replying to close with Frame { header: FrameHeader { is_final: true, rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close), mask: None}, payload: [] }
gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
gst_plugin_webrtc_signalling::server: connection closed: None this_id=192ed7ff-3b9d-45c5-be66-872cbe67d190
remove_peer{peer_id="192ed7ff-3b9d-45c5-be66-872cbe67d190"}: gst_plugin_webrtc_signalling::server: close time.busy=22.7µs time.idle=37.4µs
tungstenite::protocol: Sending pong/close
```

We now get the notification `Sending pong/close` (the closing handshake) instead
of `websocket start_send error` from step 6 with previous variant.

The `Connection reset without closing handshake` was not observed after this
change.

[send task]: 63b568f4a0/net/webrtc/signalling/src/server/mod.rs (L165)
[`WebSocket::close()` doc]: https://docs.rs/tungstenite/0.21.0/tungstenite/protocol/struct.WebSocket.html#method.close
[returns `SendAfterClosing`]: 85463b264e/src/protocol/mod.rs (L437)
[call to `do_close()`]: 85463b264e/src/protocol/mod.rs (L601)
[`close()` ensures the ws is active]: 85463b264e/src/protocol/mod.rs (L531)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
2024-02-01 15:08:11 +05:30